Author: kmoore
Date: Wed Jul 31 22:11:59 2013
New Revision: 395927
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=395927
Log:
Pull in changes up to trunk
Removed:
team/kmoore/cel_backend_refactor/res/res_pjsip_endpoint_identifier_constant.c
Modified:
team/kmoore/cel_backend_refactor/ (props changed)
team/kmoore/cel_backend_refactor/CHANGES
team/kmoore/cel_backend_refactor/bridges/bridge_native_rtp.c
team/kmoore/cel_backend_refactor/channels/chan_pjsip.c
team/kmoore/cel_backend_refactor/channels/chan_sip.c
team/kmoore/cel_backend_refactor/configs/pjsip.conf.sample
team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip.h
team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_exten_state.h
team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_pubsub.h
team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_session.h
team/kmoore/cel_backend_refactor/res/res_agi.c
team/kmoore/cel_backend_refactor/res/res_pjsip/include/res_pjsip_private.h
team/kmoore/cel_backend_refactor/res/res_pjsip/pjsip_options.c
team/kmoore/cel_backend_refactor/tests/test_cel.c
Propchange: team/kmoore/cel_backend_refactor/
------------------------------------------------------------------------------
--- cel_integrated (original)
+++ cel_integrated Wed Jul 31 22:11:59 2013
@@ -1,1 +1,1 @@
-/team/kmoore/event_system_strip:1-395850
+/team/kmoore/event_system_strip:1-395926
Modified: team/kmoore/cel_backend_refactor/CHANGES
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/CHANGES?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/CHANGES (original)
+++ team/kmoore/cel_backend_refactor/CHANGES Wed Jul 31 22:11:59 2013
@@ -304,6 +304,10 @@
* The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
and AsyncAGIEnd.
+
+ * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
+ will start the playback of the audio at the position specified. It will
+ also return the final position of the file in 'endpos'.
CDR (Call Detail Records)
------------------
Modified: team/kmoore/cel_backend_refactor/bridges/bridge_native_rtp.c
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/bridges/bridge_native_rtp.c?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/bridges/bridge_native_rtp.c (original)
+++ team/kmoore/cel_backend_refactor/bridges/bridge_native_rtp.c Wed Jul 31
22:11:59 2013
@@ -46,46 +46,11 @@
#include "asterisk/frame.h"
#include "asterisk/rtp_engine.h"
-/*! \brief Forward declarations for frame hook usage */
-static int native_rtp_bridge_join(struct ast_bridge *bridge, struct
ast_bridge_channel *bridge_channel);
-static void native_rtp_bridge_leave(struct ast_bridge *bridge, struct
ast_bridge_channel *bridge_channel);
-
/*! \brief Internal structure which contains information about bridged RTP
channels */
struct native_rtp_bridge_data {
/*! \brief Framehook used to intercept certain control frames */
int id;
};
-
-/*! \brief Frame hook that is called to intercept hold/unhold */
-static struct ast_frame *native_rtp_framehook(struct ast_channel *chan, struct
ast_frame *f, enum ast_framehook_event event, void *data)
-{
- RAII_VAR(struct ast_bridge *, bridge, NULL, ao2_cleanup);
-
- if (!f || (event != AST_FRAMEHOOK_EVENT_WRITE)) {
- return f;
- }
-
- ast_channel_lock(chan);
- bridge = ast_channel_get_bridge(chan);
- ast_channel_unlock(chan);
-
- /* It's safe for NULL to be passed to both of these, bridge_channel
isn't used at all */
- if (bridge) {
- if (f->subclass.integer == AST_CONTROL_HOLD) {
-
native_rtp_bridge_leave(ast_channel_internal_bridge(chan), NULL);
- } else if ((f->subclass.integer == AST_CONTROL_UNHOLD) ||
(f->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) {
-
native_rtp_bridge_join(ast_channel_internal_bridge(chan), NULL);
- }
- }
-
- return f;
-}
-
-/*! \brief Internal helper function which checks whether the channels are
compatible with our native bridging */
-static int native_rtp_bridge_capable(struct ast_channel *chan)
-{
- return !ast_channel_has_audio_frame_or_monitor(chan);
-}
/*! \brief Internal helper function which gets all RTP information (glue and
instances) relating to the given channels */
static enum ast_rtp_glue_result native_rtp_bridge_get(struct ast_channel *c0,
struct ast_channel *c1, struct ast_rtp_glue **glue0,
@@ -140,6 +105,130 @@
}
return audio_glue0_res;
+}
+
+/*! \brief Start RTP native bridging */
+static int native_rtp_bridge_start(struct ast_bridge *bridge)
+{
+ struct ast_bridge_channel *c0 = AST_LIST_FIRST(&bridge->channels);
+ struct ast_bridge_channel *c1 = AST_LIST_LAST(&bridge->channels);
+ enum ast_rtp_glue_result native_type;
+ struct ast_rtp_glue *glue0, *glue1;
+ struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
*vinstance0 = NULL;
+ struct ast_rtp_instance *vinstance1 = NULL, *tinstance0 = NULL,
*tinstance1 = NULL;
+ RAII_VAR(struct ast_format_cap *, cap0, ast_format_cap_alloc_nolock(),
ast_format_cap_destroy);
+ RAII_VAR(struct ast_format_cap *, cap1, ast_format_cap_alloc_nolock(),
ast_format_cap_destroy);
+
+ if (c0 == c1) {
+ return 0;
+ }
+
+ native_type = native_rtp_bridge_get(c0->chan, c1->chan, &glue0, &glue1,
&instance0, &instance1, &vinstance0, &vinstance1);
+
+ if (glue0->get_codec) {
+ glue0->get_codec(c0->chan, cap0);
+ }
+ if (glue1->get_codec) {
+ glue1->get_codec(c1->chan, cap1);
+ }
+
+ switch (native_type) {
+ case AST_RTP_GLUE_RESULT_LOCAL:
+ if (ast_rtp_instance_get_engine(instance0)->local_bridge) {
+
ast_rtp_instance_get_engine(instance0)->local_bridge(instance0, instance1);
+ }
+ if (ast_rtp_instance_get_engine(instance1)->local_bridge) {
+
ast_rtp_instance_get_engine(instance1)->local_bridge(instance1, instance0);
+ }
+ ast_rtp_instance_set_bridged(instance0, instance1);
+ ast_rtp_instance_set_bridged(instance1, instance0);
+ ast_debug(2, "Locally RTP bridged '%s' and '%s' in stack\n",
+ ast_channel_name(c0->chan), ast_channel_name(c1->chan));
+ break;
+
+ case AST_RTP_GLUE_RESULT_REMOTE:
+ glue0->update_peer(c0->chan, instance1, vinstance1, tinstance1,
cap1, 0);
+ glue1->update_peer(c1->chan, instance0, vinstance0, tinstance0,
cap0, 0);
+ ast_debug(2, "Remotely bridged '%s' and '%s' - media will flow
directly between them\n",
+ ast_channel_name(c0->chan), ast_channel_name(c1->chan));
+ break;
+ case AST_RTP_GLUE_RESULT_FORBID:
+ break;
+ }
+
+ return 0;
+}
+
+/*! \brief Stop RTP native bridging */
+static void native_rtp_bridge_stop(struct ast_bridge *bridge)
+{
+ struct ast_bridge_channel *c0 = AST_LIST_FIRST(&bridge->channels);
+ struct ast_bridge_channel *c1 = AST_LIST_LAST(&bridge->channels);
+ enum ast_rtp_glue_result native_type;
+ struct ast_rtp_glue *glue0, *glue1 = NULL;
+ struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
*vinstance0 = NULL, *vinstance1 = NULL;
+
+ if (c0 == c1) {
+ return;
+ }
+
+ native_type = native_rtp_bridge_get(c0->chan, c1 ? c1->chan : NULL,
&glue0, &glue1, &instance0, &instance1, &vinstance0, &vinstance1);
+
+ switch (native_type) {
+ case AST_RTP_GLUE_RESULT_LOCAL:
+ if (ast_rtp_instance_get_engine(instance0)->local_bridge) {
+
ast_rtp_instance_get_engine(instance0)->local_bridge(instance0, NULL);
+ }
+ if (instance1 &&
ast_rtp_instance_get_engine(instance1)->local_bridge) {
+
ast_rtp_instance_get_engine(instance1)->local_bridge(instance1, NULL);
+ }
+ ast_rtp_instance_set_bridged(instance0, NULL);
+ if (instance1) {
+ ast_rtp_instance_set_bridged(instance1, NULL);
+ }
+ break;
+ case AST_RTP_GLUE_RESULT_REMOTE:
+ glue0->update_peer(c0->chan, NULL, NULL, NULL, NULL, 0);
+ if (glue1) {
+ glue1->update_peer(c1->chan, NULL, NULL, NULL, NULL, 0);
+ }
+ break;
+ case AST_RTP_GLUE_RESULT_FORBID:
+ break;
+ }
+
+ ast_debug(2, "Discontinued RTP bridging of '%s' and '%s' - media will
flow through Asterisk core\n",
+ ast_channel_name(c0->chan), ast_channel_name(c1->chan));
+}
+
+/*! \brief Frame hook that is called to intercept hold/unhold */
+static struct ast_frame *native_rtp_framehook(struct ast_channel *chan, struct
ast_frame *f, enum ast_framehook_event event, void *data)
+{
+ RAII_VAR(struct ast_bridge *, bridge, NULL, ao2_cleanup);
+
+ if (!f || (event != AST_FRAMEHOOK_EVENT_WRITE)) {
+ return f;
+ }
+
+ ast_channel_lock(chan);
+ bridge = ast_channel_get_bridge(chan);
+ ast_channel_unlock(chan);
+
+ if (bridge) {
+ if (f->subclass.integer == AST_CONTROL_HOLD) {
+ native_rtp_bridge_stop(bridge);
+ } else if ((f->subclass.integer == AST_CONTROL_UNHOLD) ||
(f->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) {
+ native_rtp_bridge_start(bridge);
+ }
+ }
+
+ return f;
+}
+
+/*! \brief Internal helper function which checks whether the channels are
compatible with our native bridging */
+static int native_rtp_bridge_capable(struct ast_channel *chan)
+{
+ return !ast_channel_has_audio_frame_or_monitor(chan);
}
static int native_rtp_bridge_compatible(struct ast_bridge *bridge)
@@ -254,15 +343,7 @@
return -1;
}
-/*
- * BUGBUG The RTP native bridge technology should use tech_pvt not bridge_pvt.
- *
- * This technology needs to be reworked to not change the
- * tech_pvt of channels other than the one that is currently
- * entering/leaving before it can actually use the correct
- * pointer.
- */
- bridge_channel->bridge_pvt = data;
+ bridge_channel->tech_pvt = data;
return 0;
}
@@ -270,7 +351,7 @@
/*! \brief Helper function which removes frame hook from bridge channel */
static void native_rtp_bridge_framehook_detach(struct ast_bridge_channel
*bridge_channel)
{
- RAII_VAR(struct native_rtp_bridge_data *, data,
bridge_channel->bridge_pvt, ao2_cleanup);
+ RAII_VAR(struct native_rtp_bridge_data *, data,
bridge_channel->tech_pvt, ao2_cleanup);
if (!data) {
return;
@@ -279,65 +360,17 @@
ast_channel_lock(bridge_channel->chan);
ast_framehook_detach(bridge_channel->chan, data->id);
ast_channel_unlock(bridge_channel->chan);
- bridge_channel->bridge_pvt = NULL;
+ bridge_channel->tech_pvt = NULL;
}
static int native_rtp_bridge_join(struct ast_bridge *bridge, struct
ast_bridge_channel *bridge_channel)
{
- struct ast_bridge_channel *c0 = AST_LIST_FIRST(&bridge->channels);
- struct ast_bridge_channel *c1 = AST_LIST_LAST(&bridge->channels);
- enum ast_rtp_glue_result native_type;
- struct ast_rtp_glue *glue0, *glue1;
- struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
*vinstance0 = NULL;
- struct ast_rtp_instance *vinstance1 = NULL, *tinstance0 = NULL,
*tinstance1 = NULL;
- RAII_VAR(struct ast_format_cap *, cap0, ast_format_cap_alloc_nolock(),
ast_format_cap_destroy);
- RAII_VAR(struct ast_format_cap *, cap1, ast_format_cap_alloc_nolock(),
ast_format_cap_destroy);
-
- native_rtp_bridge_framehook_detach(c0);
- if (native_rtp_bridge_framehook_attach(c0)) {
+ native_rtp_bridge_framehook_detach(bridge_channel);
+ if (native_rtp_bridge_framehook_attach(bridge_channel)) {
return -1;
}
- native_rtp_bridge_framehook_detach(c1);
- if (native_rtp_bridge_framehook_attach(c1)) {
- native_rtp_bridge_framehook_detach(c0);
- return -1;
- }
-
- native_type = native_rtp_bridge_get(c0->chan, c1->chan, &glue0, &glue1,
&instance0, &instance1, &vinstance0, &vinstance1);
-
- if (glue0->get_codec) {
- glue0->get_codec(c0->chan, cap0);
- }
- if (glue1->get_codec) {
- glue1->get_codec(c1->chan, cap1);
- }
-
- switch (native_type) {
- case AST_RTP_GLUE_RESULT_LOCAL:
- if (ast_rtp_instance_get_engine(instance0)->local_bridge) {
-
ast_rtp_instance_get_engine(instance0)->local_bridge(instance0, instance1);
- }
- if (ast_rtp_instance_get_engine(instance1)->local_bridge) {
-
ast_rtp_instance_get_engine(instance1)->local_bridge(instance1, instance0);
- }
- ast_rtp_instance_set_bridged(instance0, instance1);
- ast_rtp_instance_set_bridged(instance1, instance0);
- ast_debug(2, "Locally RTP bridged '%s' and '%s' in stack\n",
- ast_channel_name(c0->chan), ast_channel_name(c1->chan));
- break;
-
- case AST_RTP_GLUE_RESULT_REMOTE:
- glue0->update_peer(c0->chan, instance1, vinstance1, tinstance1,
cap1, 0);
- glue1->update_peer(c1->chan, instance0, vinstance0, tinstance0,
cap0, 0);
- ast_debug(2, "Remotely bridged '%s' and '%s' - media will flow
directly between them\n",
- ast_channel_name(c0->chan), ast_channel_name(c1->chan));
- break;
- case AST_RTP_GLUE_RESULT_FORBID:
- break;
- }
-
- return 0;
+ return native_rtp_bridge_start(bridge);
}
static void native_rtp_bridge_unsuspend(struct ast_bridge *bridge, struct
ast_bridge_channel *bridge_channel)
@@ -347,44 +380,9 @@
static void native_rtp_bridge_leave(struct ast_bridge *bridge, struct
ast_bridge_channel *bridge_channel)
{
- struct ast_bridge_channel *c0 = AST_LIST_FIRST(&bridge->channels) ?
AST_LIST_FIRST(&bridge->channels) : bridge_channel;
- struct ast_bridge_channel *c1 = AST_LIST_LAST(&bridge->channels);
- enum ast_rtp_glue_result native_type;
- struct ast_rtp_glue *glue0, *glue1 = NULL;
- struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
*vinstance0 = NULL, *vinstance1 = NULL;
-
- native_rtp_bridge_framehook_detach(c0);
- if (c1) {
- native_rtp_bridge_framehook_detach(c1);
- }
-
- native_type = native_rtp_bridge_get(c0->chan, c1 ? c1->chan : NULL,
&glue0, &glue1, &instance0, &instance1, &vinstance0, &vinstance1);
-
- switch (native_type) {
- case AST_RTP_GLUE_RESULT_LOCAL:
- if (ast_rtp_instance_get_engine(instance0)->local_bridge) {
-
ast_rtp_instance_get_engine(instance0)->local_bridge(instance0, NULL);
- }
- if (instance1 &&
ast_rtp_instance_get_engine(instance1)->local_bridge) {
-
ast_rtp_instance_get_engine(instance1)->local_bridge(instance1, NULL);
- }
- ast_rtp_instance_set_bridged(instance0, instance1);
- if (instance1) {
- ast_rtp_instance_set_bridged(instance1, instance0);
- }
- break;
- case AST_RTP_GLUE_RESULT_REMOTE:
- glue0->update_peer(c0->chan, NULL, NULL, NULL, NULL, 0);
- if (glue1) {
- glue1->update_peer(c1->chan, NULL, NULL, NULL, NULL, 0);
- }
- break;
- case AST_RTP_GLUE_RESULT_FORBID:
- break;
- }
-
- ast_debug(2, "Discontinued RTP bridging of '%s' and '%s' - media will
flow through Asterisk core\n",
- ast_channel_name(c0->chan), ast_channel_name(c1->chan));
+ native_rtp_bridge_framehook_detach(bridge_channel);
+
+ native_rtp_bridge_stop(bridge);
}
static int native_rtp_bridge_write(struct ast_bridge *bridge, struct
ast_bridge_channel *bridge_channel, struct ast_frame *frame)
Modified: team/kmoore/cel_backend_refactor/channels/chan_pjsip.c
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/channels/chan_pjsip.c?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/channels/chan_pjsip.c (original)
+++ team/kmoore/cel_backend_refactor/channels/chan_pjsip.c Wed Jul 31 22:11:59
2013
@@ -577,7 +577,7 @@
return NULL;
}
- /* If res_sip_session is ever updated to create/destroy
ast_sip_session_media
+ /* If res_pjsip_session is ever updated to create/destroy
ast_sip_session_media
* during a call such as if multiple same-type stream support is
introduced,
* these will need to be recaptured as well */
pvt->media[SIP_MEDIA_AUDIO] = ao2_find(session->media, "audio",
OBJ_KEY);
Modified: team/kmoore/cel_backend_refactor/channels/chan_sip.c
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/channels/chan_sip.c?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/channels/chan_sip.c (original)
+++ team/kmoore/cel_backend_refactor/channels/chan_sip.c Wed Jul 31 22:11:59
2013
@@ -23666,6 +23666,7 @@
ast_log(LOG_WARNING, "Forbidden - wrong password on
authentication for REGISTER for '%s' to '%s'\n", p->registry->username,
p->registry->hostname);
AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg
ptr unref from handle_response_register 403"));
r->regstate = REG_STATE_NOAUTH;
+ sip_publish_registry(r->username, r->hostname,
regstate2str(r->regstate));
pvt_set_needdestroy(p, "received 403 response");
break;
case 404: /* Not found */
@@ -23674,6 +23675,7 @@
if (r->call)
r->call = dialog_unref(r->call, "unsetting
registry->call pointer-- case 404");
r->regstate = REG_STATE_REJECTED;
+ sip_publish_registry(r->username, r->hostname,
regstate2str(r->regstate));
AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg
ptr unref from handle_response_register 404"));
break;
case 407: /* Proxy auth */
@@ -23714,6 +23716,7 @@
if (r->call)
r->call = dialog_unref(r->call, "unsetting
registry->call pointer-- case 479");
r->regstate = REG_STATE_REJECTED;
+ sip_publish_registry(r->username, r->hostname,
regstate2str(r->regstate));
AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg
ptr unref from handle_response_register 479"));
break;
case 200: /* 200 OK */
Modified: team/kmoore/cel_backend_refactor/configs/pjsip.conf.sample
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/configs/pjsip.conf.sample?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/configs/pjsip.conf.sample (original)
+++ team/kmoore/cel_backend_refactor/configs/pjsip.conf.sample Wed Jul 31
22:11:59 2013
@@ -1,4 +1,4 @@
-; This is an in-flux configuration file for the res_sip module, it will change
as things progress
+; This is an in-flux configuration file for the res_pjsip module, it will
change as things progress
;;; Transports
;[local]
Modified: team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip.h
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip.h?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip.h (original)
+++ team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip.h Wed Jul 31
22:11:59 2013
@@ -16,8 +16,8 @@
* at the top of the source tree.
*/
-#ifndef _RES_SIP_H
-#define _RES_SIP_H
+#ifndef _RES_PJSIP_H
+#define _RES_PJSIP_H
#include "asterisk/stringfields.h"
/* Needed for struct ast_sockaddr */
@@ -1499,4 +1499,4 @@
int ast_sip_initialize_sorcery_global(struct ast_sorcery *sorcery);
-#endif /* _RES_SIP_H */
+#endif /* _RES_PJSIP_H */
Modified:
team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_exten_state.h
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_exten_state.h?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_exten_state.h
(original)
+++ team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_exten_state.h
Wed Jul 31 22:11:59 2013
@@ -16,8 +16,8 @@
* at the top of the source tree.
*/
-#ifndef _RES_SIP_EXTEN_STATE_H
-#define _RES_SIP_EXTEN_STATE_H
+#ifndef _RES_PJSIP_EXTEN_STATE_H
+#define _RES_PJSIP_EXTEN_STATE_H
#include "asterisk/stringfields.h"
#include "asterisk/linkedlists.h"
@@ -91,4 +91,4 @@
*/
void ast_sip_unregister_exten_state_provider(struct
ast_sip_exten_state_provider *obj);
-#endif
+#endif /* _RES_PJSIP_EXTEN_STATE_H */
Modified: team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_pubsub.h
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_pubsub.h?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_pubsub.h
(original)
+++ team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_pubsub.h Wed
Jul 31 22:11:59 2013
@@ -16,8 +16,8 @@
* at the top of the source tree.
*/
-#ifndef _RES_SIP_PUBSUB_H
-#define _RES_SIP_PUBSUB_H
+#ifndef _RES_PJSIP_PUBSUB_H
+#define _RES_PJSIP_PUBSUB_H
#include "asterisk/linkedlists.h"
@@ -513,4 +513,4 @@
*/
void ast_sip_unregister_subscription_handler(struct
ast_sip_subscription_handler *handler);
-#endif /* RES_SIP_PUBSUB_H */
+#endif /* RES_PJSIP_PUBSUB_H */
Modified: team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_session.h
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_session.h?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_session.h
(original)
+++ team/kmoore/cel_backend_refactor/include/asterisk/res_pjsip_session.h Wed
Jul 31 22:11:59 2013
@@ -16,8 +16,8 @@
* at the top of the source tree.
*/
-#ifndef _RES_SIP_SESSION_H
-#define _RES_SIP_SESSION_H
+#ifndef _RES_PJSIP_SESSION_H
+#define _RES_PJSIP_SESSION_H
/* Needed for pj_timer_entry definition */
#include "pjlib.h"
@@ -141,7 +141,7 @@
/*! Top priority. Supplements with this priority are those that need to
run before any others */
AST_SIP_SESSION_SUPPLEMENT_PRIORITY_FIRST = 0,
/*! Channel creation priority.
- * chan_gulp creates a channel at this priority. If your supplement
depends on being run before
+ * chan_pjsip creates a channel at this priority. If your supplement
depends on being run before
* or after channel creation, then set your priority to be lower or
higher than this value.
*/
AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL = 1000000,
@@ -466,7 +466,7 @@
* \brief Send a reinvite or UPDATE on a session
*
* This method will inspect the session in order to construct an appropriate
- * session refresh request. As with any outgoing request in res_sip_session,
+ * session refresh request. As with any outgoing request in res_pjsip_session,
* this will call into registered supplements in case they wish to add
anything.
*
* Note: The on_request_creation callback may or may not be called in the same
@@ -558,4 +558,4 @@
*/
void ast_sip_session_resume_reinvite(struct ast_sip_session *session);
-#endif /* _RES_SIP_SESSION_H */
+#endif /* _RES_PJSIP_SESSION_H */
Modified: team/kmoore/cel_backend_refactor/res/res_agi.c
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/res/res_agi.c?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/res/res_agi.c (original)
+++ team/kmoore/cel_backend_refactor/res/res_agi.c Wed Jul 31 22:11:59 2013
@@ -155,13 +155,19 @@
<para>Defaults to <literal>#</literal></para>
</parameter>
<parameter name="pausechr" />
+ <parameter name="offsetms">
+ <para>Offset, in milliseconds, to start the
audio playback</para>
+ </parameter>
</syntax>
<description>
<para>Send the given file, allowing playback to be
controlled by the given
digits, if any. Use double quotes for the digits if you
wish none to be
- permitted. Returns <literal>0</literal> if playback
completes without a digit
+ permitted. If offsetms is provided then the audio will
seek to offsetms
+ before play starts. Returns <literal>0</literal> if
playback completes without a digit
being pressed, or the ASCII numerical value of the
digit if one was pressed,
- or <literal>-1</literal> on error or if the channel was
disconnected.</para>
+ or <literal>-1</literal> on error or if the channel was
disconnected. Returns the
+ position where playback was terminated as endpos.</para>
+
<para>It sets the following channel variables upon
completion:</para>
<variablelist>
<variable name="CPLAYBACKSTATUS">
@@ -368,9 +374,9 @@
</parameter>
</syntax>
<description>
- <para>Receives a string of text on a channel. Most
channels
+ <para>Receives a string of text on a channel. Most
channels
do not support the reception of text. Returns
<literal>-1</literal> for failure
- or <literal>1</literal> for success, and the string in
parenthesis.</para>
+ or <literal>1</literal> for success, and the string in
parenthesis.</para>
</description>
</agi>
<agi name="record file" language="en_US">
@@ -2092,7 +2098,7 @@
long offsetms = 0;
char offsetbuf[20];
- if (argc < 5 || argc > 9) {
+ if (argc < 5 || argc > 10) {
return RESULT_SHOWUSAGE;
}
@@ -2116,7 +2122,11 @@
suspend = argv[8];
}
- res = ast_control_streamfile(chan, argv[3], fwd, rev, stop, suspend,
NULL, skipms, NULL);
+ if (argc > 9 && (sscanf(argv[9], "%30ld", &offsetms) != 1)) {
+ return RESULT_SHOWUSAGE;
+ }
+
+ res = ast_control_streamfile(chan, argv[3], fwd, rev, stop, suspend,
NULL, skipms, &offsetms);
/* If we stopped on one of our stop keys, return 0 */
if (res > 0 && stop && strchr(stop, res)) {
@@ -2137,7 +2147,7 @@
snprintf(offsetbuf, sizeof(offsetbuf), "%ld", offsetms);
pbx_builtin_setvar_helper(chan, "CPLAYBACKOFFSET", offsetbuf);
- ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
+ ast_agi_send(agi->fd, chan, "200 result=%d endpos=%ld\n", res,
offsetms);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
@@ -2518,7 +2528,7 @@
}
ast_dsp_set_threshold(sildet,
ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE));
}
-
+
/* backward compatibility, if no offset given, arg[6] would have been
* caught below and taken to be a beep, else if it is a digit then it
is a
* offset */
@@ -2881,7 +2891,7 @@
break;
}
} while (1);
-
+
if (res)
ast_agi_send(agi->fd, chan, "200 result=0\n");
else
@@ -3294,15 +3304,15 @@
{ { "noop", NULL }, handle_noop, NULL, NULL, 1 },
{ { "receive", "char", NULL }, handle_recvchar, NULL, NULL, 0 },
{ { "receive", "text", NULL }, handle_recvtext, NULL, NULL, 0 },
- { { "record", "file", NULL }, handle_recordfile, NULL, NULL, 0 },
+ { { "record", "file", NULL }, handle_recordfile, NULL, NULL, 0 },
{ { "say", "alpha", NULL }, handle_sayalpha, NULL, NULL, 0},
{ { "say", "digits", NULL }, handle_saydigits, NULL, NULL, 0 },
{ { "say", "number", NULL }, handle_saynumber, NULL, NULL, 0 },
- { { "say", "phonetic", NULL }, handle_sayphonetic, NULL, NULL, 0},
- { { "say", "date", NULL }, handle_saydate, NULL, NULL, 0},
- { { "say", "time", NULL }, handle_saytime, NULL, NULL, 0},
+ { { "say", "phonetic", NULL }, handle_sayphonetic, NULL, NULL, 0},
+ { { "say", "date", NULL }, handle_saydate, NULL, NULL, 0},
+ { { "say", "time", NULL }, handle_saytime, NULL, NULL, 0},
{ { "say", "datetime", NULL }, handle_saydatetime, NULL, NULL, 0},
- { { "send", "image", NULL }, handle_sendimage, NULL, NULL, 0},
+ { { "send", "image", NULL }, handle_sendimage, NULL, NULL, 0},
{ { "send", "text", NULL }, handle_sendtext, NULL, NULL, 0},
{ { "set", "autohangup", NULL }, handle_autohangup, NULL, NULL, 0},
{ { "set", "callerid", NULL }, handle_setcallerid, NULL, NULL, 0},
@@ -3706,7 +3716,7 @@
const char *sighup_str;
const char *exit_on_hangup_str;
int exit_on_hangup;
-
+
ast_channel_lock(chan);
sighup_str = pbx_builtin_getvar_helper(chan, "AGISIGHUP");
send_sighup = !ast_false(sighup_str);
@@ -3721,7 +3731,7 @@
close(agi->ctrl);
return AGI_RESULT_FAILURE;
}
-
+
setlinebuf(readf);
setup_env(chan, request, agi->fd, (agi->audio > -1), argc, argv);
for (;;) {
Modified:
team/kmoore/cel_backend_refactor/res/res_pjsip/include/res_pjsip_private.h
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/res/res_pjsip/include/res_pjsip_private.h?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/res/res_pjsip/include/res_pjsip_private.h
(original)
+++ team/kmoore/cel_backend_refactor/res/res_pjsip/include/res_pjsip_private.h
Wed Jul 31 22:11:59 2013
@@ -5,8 +5,8 @@
* Author: mjordan
*/
-#ifndef RES_SIP_PRIVATE_H_
-#define RES_SIP_PRIVATE_H_
+#ifndef RES_PJSIP_PRIVATE_H_
+#define RES_PJSIP_PRIVATE_H_
struct ao2_container;
@@ -71,4 +71,4 @@
*/
int ast_sip_initialize_global(void);
-#endif /* RES_SIP_PRIVATE_H_ */
+#endif /* RES_PJSIP_PRIVATE_H_ */
Modified: team/kmoore/cel_backend_refactor/res/res_pjsip/pjsip_options.c
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/res/res_pjsip/pjsip_options.c?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/res/res_pjsip/pjsip_options.c (original)
+++ team/kmoore/cel_backend_refactor/res/res_pjsip/pjsip_options.c Wed Jul 31
22:11:59 2013
@@ -625,9 +625,9 @@
switch (cmd) {
case CLI_INIT:
- e->command = "sip qualify";
+ e->command = "pjsip qualify";
e->usage =
- "Usage: sip qualify <endpoint>\n"
+ "Usage: pjsip qualify <endpoint>\n"
" Send a SIP OPTIONS request to all contacts on
the endpoint.\n";
return NULL;
case CLI_GENERATE:
@@ -653,7 +653,7 @@
}
static struct ast_cli_entry cli_options[] = {
- AST_CLI_DEFINE(cli_qualify, "Send an OPTIONS request to a SIP endpoint")
+ AST_CLI_DEFINE(cli_qualify, "Send an OPTIONS request to a PJSIP
endpoint")
};
static int sched_qualifies_hash_fn(const void *obj, int flags)
Modified: team/kmoore/cel_backend_refactor/tests/test_cel.c
URL:
http://svnview.digium.com/svn/asterisk/team/kmoore/cel_backend_refactor/tests/test_cel.c?view=diff&rev=395927&r1=395926&r2=395927
==============================================================================
--- team/kmoore/cel_backend_refactor/tests/test_cel.c (original)
+++ team/kmoore/cel_backend_refactor/tests/test_cel.c Wed Jul 31 22:11:59 2013
@@ -561,6 +561,10 @@
return AST_TEST_PASS;
}
+/* XXX Validation needs to be reworked on a per-channel basis before
+ * test_cel_single_multiparty_bridge and test_cel_dial_answer_multiparty
+ * can operate properly. */
+#if 0
AST_TEST_DEFINE(test_cel_single_multiparty_bridge)
{
RAII_VAR(struct ast_channel *, chan_alice, NULL, safe_channel_release);
@@ -623,6 +627,7 @@
return AST_TEST_PASS;
}
+#endif
#define EMULATE_DIAL(channel, dialstring) do { \
EMULATE_APP_DATA(channel, 1, "Dial", dialstring); \
@@ -1014,6 +1019,7 @@
return AST_TEST_PASS;
}
+#if 0
AST_TEST_DEFINE(test_cel_dial_answer_multiparty)
{
RAII_VAR(struct ast_channel *, chan_alice, NULL, safe_channel_release);
@@ -1097,6 +1103,7 @@
return AST_TEST_PASS;
}
+#endif
AST_TEST_DEFINE(test_cel_blind_transfer)
{
@@ -1891,7 +1898,7 @@
AST_TEST_UNREGISTER(test_cel_single_bridge_continue);
AST_TEST_UNREGISTER(test_cel_single_twoparty_bridge_a);
AST_TEST_UNREGISTER(test_cel_single_twoparty_bridge_b);
- AST_TEST_UNREGISTER(test_cel_single_multiparty_bridge);
+ /*AST_TEST_UNREGISTER(test_cel_single_multiparty_bridge);*/
AST_TEST_UNREGISTER(test_cel_dial_unanswered);
AST_TEST_UNREGISTER(test_cel_dial_congestion);
@@ -1902,7 +1909,7 @@
AST_TEST_UNREGISTER(test_cel_dial_answer_no_bridge);
AST_TEST_UNREGISTER(test_cel_dial_answer_twoparty_bridge_a);
AST_TEST_UNREGISTER(test_cel_dial_answer_twoparty_bridge_b);
- AST_TEST_UNREGISTER(test_cel_dial_answer_multiparty);
+ /*AST_TEST_UNREGISTER(test_cel_dial_answer_multiparty);*/
AST_TEST_UNREGISTER(test_cel_blind_transfer);
AST_TEST_UNREGISTER(test_cel_attended_transfer_bridges_swap);
@@ -1964,7 +1971,7 @@
AST_TEST_REGISTER(test_cel_single_bridge_continue);
AST_TEST_REGISTER(test_cel_single_twoparty_bridge_a);
AST_TEST_REGISTER(test_cel_single_twoparty_bridge_b);
- AST_TEST_REGISTER(test_cel_single_multiparty_bridge);
+ /*AST_TEST_REGISTER(test_cel_single_multiparty_bridge);*/
AST_TEST_REGISTER(test_cel_dial_unanswered);
AST_TEST_REGISTER(test_cel_dial_congestion);
@@ -1975,7 +1982,7 @@
AST_TEST_REGISTER(test_cel_dial_answer_no_bridge);
AST_TEST_REGISTER(test_cel_dial_answer_twoparty_bridge_a);
AST_TEST_REGISTER(test_cel_dial_answer_twoparty_bridge_b);
- AST_TEST_REGISTER(test_cel_dial_answer_multiparty);
+ /*AST_TEST_REGISTER(test_cel_dial_answer_multiparty);*/
AST_TEST_REGISTER(test_cel_blind_transfer);
AST_TEST_REGISTER(test_cel_attended_transfer_bridges_swap);
--
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